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Audio generation has achieved remarkable progress with the advance of sophisticated generative models, such as diffusion models (DMs) and autoregressive (AR) models. However, due to the naturally significant sequence length of audio, the…

声音 · 计算机科学 2024-12-18 Kai Qiu , Xiang Li , Hao Chen , Jie Sun , Jinglu Wang , Zhe Lin , Marios Savvides , Bhiksha Raj

Machine Speech Chain, simulating the human perception-production loop, proves effective in jointly improving ASR and TTS. We propose TokenChain, a fully discrete speech chain coupling semantic-token ASR with a two-stage TTS: an…

音频与语音处理 · 电气工程与系统科学 2026-05-05 Mingxuan Wang , Satoshi Nakamura

Multi-step reasoning ability is fundamental to many natural language tasks, yet it is unclear what constitutes a good reasoning chain and how to evaluate them. Most existing methods focus solely on whether the reasoning chain leads to the…

计算与语言 · 计算机科学 2023-12-04 Archiki Prasad , Swarnadeep Saha , Xiang Zhou , Mohit Bansal

In reasoning chains generated by large language models (LLMs), initial errors often propagate and undermine the reliability of the final conclusion. Current LLM-based error detection methods often fail to detect propagated errors because…

机器学习 · 计算机科学 2025-09-30 Weiqiu You , Anton Xue , Shreya Havaldar , Delip Rao , Helen Jin , Chris Callison-Burch , Eric Wong

Recent work has shown that it is possible to train a single model to perform joint acoustic echo cancellation (AEC), speech enhancement, and voice separation, thereby serving as a unified frontend for robust automatic speech recognition…

音频与语音处理 · 电气工程与系统科学 2022-09-15 Tom O'Malley , Arun Narayanan , Quan Wang

This paper proposes a novel automatic speech recognition (ASR) framework called Integrated Source-Channel and Attention (ISCA) that combines the advantages of traditional systems based on the noisy source-channel model (SC) and end-to-end…

音频与语音处理 · 电气工程与系统科学 2019-10-02 Qiujia Li , Chao Zhang , Philip C. Woodland

There is increasingly more evidence that automatic speech recognition (ASR) systems are biased against different speakers and speaker groups, e.g., due to gender, age, or accent. Research on bias in ASR has so far primarily focused on…

计算与语言 · 计算机科学 2025-07-09 Tanvina Patel , Wiebke Hutiri , Aaron Yi Ding , Odette Scharenborg

The conversion from text to speech relies on the accurate mapping from linguistic to acoustic symbol sequences, for which current practice employs recurrent statistical models like recurrent neural networks. Despite the good performance of…

声音 · 计算机科学 2018-11-07 Santiago Pascual , Antonio Bonafonte , Joan Serrà

Automatic speech recognition (ASR) systems typically rely on an external endpointer (EP) model to identify speech boundaries. In this work, we propose a method to jointly train the ASR and EP tasks in a single end-to-end (E2E) multitask…

声音 · 计算机科学 2023-02-16 Shaan Bijwadia , Shuo-yiin Chang , Bo Li , Tara Sainath , Chao Zhang , Yanzhang He

Recent autoregressive transformer-based speech enhancement (SE) methods have shown promising results by leveraging advanced semantic understanding and contextual modeling of speech. However, these approaches often rely on complex…

声音 · 计算机科学 2025-10-03 Luca A. Lanzendörfer , Frédéric Berdoz , Antonis Asonitis , Roger Wattenhofer

The performance of voice-controlled systems is usually influenced by accented speech. To make these systems more robust, the frontend accent recognition (AR) technologies have received increased attention in recent years. As accent is a…

音频与语音处理 · 电气工程与系统科学 2021-05-06 Zhan Zhang , Xi Chen , Yuehai Wang , Jianyi Yang

The issue of fairness arises when the automatic speech recognition (ASR) systems do not perform equally well for all subgroups of the population. In any fairness measurement studies for ASR, the open questions of how to control the nuisance…

机器学习 · 统计学 2021-09-21 Zhe Liu , Irina-Elena Veliche , Fuchun Peng

We propose ARiSE, an auto-regressive algorithm for multi-channel speech enhancement. ARiSE improves existing deep neural network (DNN) based frame-online multi-channel speech enhancement models by introducing auto-regressive connections,…

音频与语音处理 · 电气工程与系统科学 2025-06-09 Pengjie Shen , Xueliang Zhang , Zhong-Qiu Wang

Despite advances in multilingual automatic speech recognition (ASR), code-switching (CS), the mixing of languages within an utterance common in daily speech, remains a severely underexplored challenge. In this paper, we introduce HiKE: the…

计算与语言 · 计算机科学 2026-01-14 Gio Paik , Yongbeom Kim , Soungmin Lee , Sangmin Ahn , Chanwoo Kim

This article develops a general detection theory for speech analysis based on time-varying autoregressive models, which themselves generalize the classical linear predictive speech analysis framework. This theory leads to a computationally…

应用统计 · 统计学 2011-08-25 Daniel Rudoy , Thomas F. Quatieri , Patrick J. Wolfe

Measuring performance of an automatic speech recognition (ASR) system without ground-truth could be beneficial in many scenarios, especially with data from unseen domains, where performance can be highly inconsistent. In conventional ASR…

计算与语言 · 计算机科学 2019-04-11 Ruizhi Li , Gregory Sell , Hynek Hermansky

Accurately finding the wrong words in the automatic speech recognition (ASR) hypothesis and recovering them well-founded is the goal of speech error correction. In this paper, we propose a non-autoregressive speech error correction method.…

计算与语言 · 计算机科学 2024-07-19 Yuchun Shu , Bo Hu , Yifeng He , Hao Shi , Longbiao Wang , Jianwu Dang

Automatic Speech Recognition (ASR) systems have become ubiquitous in everyday applications, yet significant disparities in performance across diverse demographic groups persist. In this work, we introduce the ASR-FAIRBENCH leaderboard which…

声音 · 计算机科学 2025-05-20 Anand Rai , Satyam Rahangdale , Utkarsh Anand , Animesh Mukherjee

End-to-end (E2E) automatic speech recognition (ASR) models have become standard practice for various commercial applications. However, in real-world scenarios, the long-tailed nature of word distribution often leads E2E ASR models to…

计算与语言 · 计算机科学 2024-09-11 Yi-Cheng Wang , Li-Ting Pai , Bi-Cheng Yan , Hsin-Wei Wang , Chi-Han Lin , Berlin Chen

We release the EARS (Expressive Anechoic Recordings of Speech) dataset, a high-quality speech dataset comprising 107 speakers from diverse backgrounds, totaling in 100 hours of clean, anechoic speech data. The dataset covers a large range…

音频与语音处理 · 电气工程与系统科学 2024-06-13 Julius Richter , Yi-Chiao Wu , Steven Krenn , Simon Welker , Bunlong Lay , Shinji Watanabe , Alexander Richard , Timo Gerkmann